If the jump from 16 to 24-bit depth mainly has to do with the dynamic range of sound, and isn't likely to be perceived due to the dynamic range already allowed by 16-bit, why is the jump from 8 to 16-bit sound so instantly noticeable, even on very poor speakers?

I remember listening to a bunch of 8- and 16-bit clips back when I first got a sound card on my PC.

The jump from 8 to 16 bit is immediately noticeable because the usual way to deal with quantization error, the error between the exact value of the analog signal and the digital value, is dithering which means "converting" this error into floor noise. At 8 bit, this floor noise is some 48 dB bellow the signal if no noise shaping is applied, it is easily audible, that's 30 something dB if you are listening at normal volume, ie the volume of a quiet room which is never totally quiet, thus easily audible.

Now, if you shift to 16 bit, the noise floor is 96 dB below the main signal, which means the noise doesn't even reach audibility threshold if you listen at normal volumes, and if it somehow does, the noise floor of the 16 bit file remains buried in the noise floor of the room.

I think naike was getting at the fact that a lot of DAWs and sound applications can internally process the sound at higher bit-depths than what hardware supports (32-bit for Foobar, 48-bit for Pro Tools, etc.). Even in professional studios, most hardware sound cards and DACs only support up to 24-bit.

As other posters have commented, 16-bit has a ton of dynamic range if used properly, so what is the benefit of higher bit-rates? Dynamic accuracy. Less interpolation. Even if we're talking about rock music that only uses the top 10% of available dynamic range, that 10% represents 6,553.6 values in a 16-bit system and 1,677,721.6 values in a 24-bit system. This is the same benefit seen with an increased sampling rate. Not only does an increased sampling rate result in higher representable frequencies, it results in more accuracy in the audible range due to a decreased need for interpolation because of the additional samples.

You are speaking of accuracy, but what is it? In my opinion, you are speaking of the ability if distinguish a low level signal in when it's played at the same time as a high level one, let 0dB be the highest volume played, it corresponds to a 96 dB with 16 bit.

Suppose you have a 1st sin wave playing - 6 dB (if you considered 16 bit, the signal has an amplitude of (2^16)/2=32768) and a 2nd sin wave representing a low level detail playing at -84 dB (amplitude of 4). Does it matter if the amplitude is changed from 4 to 3 or 5?

Not that much, because this secondary signal is already 78 dB softer than the main signal, Saying that there are more value in 24 bit is accurate, but it doesn't bring anything to the auditor, all those minute variation are basically inaudible. With noise shaping we can basically reproduce secondary, tertiary, quaternary... signals that are up to 90 dB softer than the main signal, anything softer would be inaudible.

The jump from 8 to 16 bit is immediately noticeable because the usual way to deal with quantization error, the error between the exact value of the analog signal and the digital value, is dithering which means "converting" this error into floor noise. At 8 bit, this floor noise is some 48 dB bellow the signal if no noise shaping is applied, it is easily audible, that's 30 something dB if you are listening at normal volume, ie the volume of a quiet room which is never totally quiet, thus easily audible.

Now, if you shift to 16 bit, the noise floor is 96 dB below the main signal, which means the noise doesn't even reach audibility threshold if you listen at normal volumes, and if it somehow does, the noise floor of the 16 bit file remains buried in the noise floor of the room.

As other posters have commented, 16-bit has a ton of dynamic range if used properly, so what is the benefit of higher bit-rates? Dynamic accuracy. Less interpolation. Even if we're talking about rock music that only uses the top 10% of available dynamic range, that 10% represents 6,553.6 values in a 16-bit system and 1,677,721.6 values in a 24-bit system. This is the same benefit seen with an increased sampling rate. Not only does an increased sampling rate result in higher representable frequencies, it results in more accuracy in the audible range due to a decreased need for interpolation because of the additional samples.

Doesn't 'dynamic accuracy' actually mean the same as less noise? And increasing the sampling rate certainly would give us higher frequencies. But it would not give us greater accuracy in reproducing lower frequencies already covered by a lower sampling rate. Nyquist theorem and all that.

Doesn't 'dynamic accuracy' actually mean the same as less noise? And increasing the sampling rate certainly would give us higher frequencies. But it would not give us greater accuracy in reproducing lower frequencies already covered by a lower sampling rate. Nyquist theorem and all that.

A digital audio a system can reconstruct frequencies that are less than half the sampling frequency. Theoretically a 44.1 KHz system can reproduce frequencies that are lower than 22.05 KKz. Increasing the sampling rate to 96 Khz, for instance, would give us the possibility to reproduce frequencies thet are less than 48 Khz.

You are speaking of accuracy, but what is it? In my opinion, you are speaking of the ability if distinguish a low level signal in when it's played at the same time as a high level one, let 0dB be the highest volume played, it corresponds to a 96 dB with 16 bit.

Suppose you have a 1st sin wave playing - 6 dB (if you considered 16 bit, the signal has an amplitude of (2^16)/2=32768) and a 2nd sin wave representing a low level detail playing at -84 dB (amplitude of 4). Does it matter if the amplitude is changed from 4 to 3 or 5?

Not that much, because this secondary signal is already 78 dB softer than the main signal, Saying that there are more value in 24 bit is accurate, but it doesn't bring anything to the auditor, all those minute variation are basically inaudible. With noise shaping we can basically reproduce secondary, tertiary, quaternary... signals that are up to 90 dB softer than the main signal, anything softer would be inaudible.

That's not necessarily what I'm getting at (though when you say "not that much," I would say that perhaps any improvement is worthwhile even if it is not perceptible to everyone). You are speaking of dynamic range. I'm saying, regardless of signal level, there are more interstitial values from one volume level to another. Higher bit-rates yield more accurate information about the original signal.

Quote:

Originally Posted by Leporello

Doesn't 'dynamic accuracy' actually mean the same as less noise? And increasing the sampling rate certainly would give us higher frequencies. But it would not give us greater accuracy in reproducing lower frequencies already covered by a lower sampling rate. Nyquist theorem and all that.

No, because I'm not speaking of dynamic range in relation to signal-to-noise ratio. Higher sampling rates do give increased accuracy in the audible range for a number of reasons:

1) More samples means more accuracy for all frequencies. This is the idea behind Sony's SACD format which only has a 1-bit depth but a sampling rate of 2.8224 MHz. They were trying to approximate vinyl with a digital format (hence the use of analog filters in the SACD spec).

2) On a technical level, higher sampling rates allow the cut-off frequency to be moved higher and further away from the audible frequency range. The frequency response must be rolled off to zero before hitting the Nyquist frequency and, as designers discovered in the '80s, it is not possible to use a brickwall filter to near-instantly cut to zero without seriously affecting the audible frequency range.

3) Hardware performs more efficiently at certain sampling rates, though this is on a case-by-case basis. Sort of like how Benchmark arrived at a sampling rate of ~110 kHz to maximize the performance of the chips they used in the DAC1.

4) Getting further away from the topic and into less objective territory, it is possible that inaudibly high frequencies can still be perceived non-aurally. I don't think the idea has been definitively proven, but it is being researched. Yes, I know this is the Sound Science forum. Be gentle!

That's not necessarily what I'm getting at (though when you say "not that much," I would say that perhaps any improvement is worthwhile even if it is not perceptible to everyone). You are speaking of dynamic range. I'm saying, regardless of signal level, there are more interstitial values from one volume level to another. Higher bit-rates yield more accurate information about the original signal.

No, because I'm not speaking of dynamic range in relation to signal-to-noise ratio. Higher sampling rates do give increased accuracy in the audible range for a number of reasons:

1) More samples means more accuracy for all frequencies. This is the idea behind Sony's SACD format which only has a 1-bit depth but a sampling rate of 2.8224 MHz. They were trying to approximate vinyl with a digital format (hence the use of analog filters in the SACD spec).

2) On a technical level, higher sampling rates allow the cut-off frequency to be moved higher and further away from the audible frequency range. The frequency response must be rolled off to zero before hitting the Nyquist frequency and, as designers discovered in the '80s, it is not possible to use a brickwall filter to near-instantly cut to zero without seriously affecting the audible frequency range.

3) Hardware performs more efficiently at certain sampling rates, though this is on a case-by-case basis. Sort of like how Benchmark arrived at a sampling rate of ~110 kHz to maximize the performance of the chips they used in the DAC1.

4) Getting further away from the topic and into less objective territory, it is possible that inaudibly high frequencies can still be perceived non-aurally. I don't think the idea has been definitively proven, but it is being researched. Yes, I know this is the Sound Science forum. Be gentle!

To summarize, more information = more accuracy. No one needs 144 dB of dynamic range, but that's not the only benefit of using 24-bit. No one needs frequency response up to 96 kHz, but that's not the only benefit of using 192 kHz. Etc.

monoethylene, http://en.wikipedia.org/wiki/Nyquist_frequency. Unless you thought I literally meant "all frequencies," like, all frequencies in existence. I meant, more samples means more accuracy for all representable frequencies relative to the sampling rate. I figured that was implied.

To summarize, more information = more accuracy. No one needs 144 dB of dynamic range, but that's not the only benefit of using 24-bit. No one needs frequency response up to 96 kHz, but that's not the only benefit of using 192 kHz. Et

To summarize, more information = more accuracy. No one needs 144 dB of dynamic range, but that's not the only benefit of using 24-bit. No one needs frequency response up to 96 kHz, but that's not the only benefit of using 192 kHz. Etc.

monoethylene, http://en.wikipedia.org/wiki/Nyquist_frequency. Unless you thought I literally meant "all frequencies," like, all frequencies in existence. I meant, more samples means more accuracy for all representable frequencies relative to the sampling rate. I figured that was implied.

Except that you're not getting more accuracy, you're copying what you do have multiple times. There's not any more there, there if you have a higher sampling rate.

That's not necessarily what I'm getting at (though when you say "not that much," I would say that perhaps any improvement is worthwhile even if it is not perceptible to everyone). You are speaking of dynamic range. I'm saying, regardless of signal level, there are more interstitial values from one volume level to another. Higher bit-rates yield more accurate information about the original signal.

But there's point when more improvement and precision is useless, would anyone care if the speedometer of their car was precise to the 1/10000000 miles/hour? Of course not, because they can't perceive that.
When your dithered and noise shaped 16 bit file has a subjective dynamic range of 120 dB, that means the smallest detail is 120 dB softer than the loudest one, isn't that interstitial value already small enough?

If the jump from 16 to 24-bit depth mainly has to do with the dynamic range of sound, and isn't likely to be perceived due to the dynamic range already allowed by 16-bit, why is the jump from 8 to 16-bit sound so instantly noticeable, even on very poor speakers?

I remember listening to a bunch of 8- and 16-bit clips back when I first got a sound card on my PC.

The difference is in the exponential growth of binary numbers and in the fact that 16-bit is near perfect while 8-bit is not. 16-bit allows a very large dynamic range with a step size so small that we can't hear it. 8-bit is so much smaller (remember its 2^8 times smaller, not half as small) that to get a reasonable dynamic range you probably need to use a much bigger step-size which sounds worse. Going up to 24-bit doesn't really matter because 16 is already good enough, there is nothing more to gain. It's like refresh rates on a TV, after about 30hz differences are imperceptible because the human optical system doesn't refresh faster than that. Or how a 2Mpixel picture looks exactly the same as a 20Mpixel picture unless you blow it up. Coincidently the advantage of having higher bit-depths is the similar to the advantage of filming well over hi-def 1080 (the red-one from oakley has like 5k resolution I think) that advantage is in processing and filtering. You can use the extra bits/pixels to fiddle with the sound and move the error into the low-order bits so no one notices. You can combine near-by pixels to get a brighter picture or you can use a small portion of the shot for the entire shot etc... In audio there are similar tricks like different filters and such.

To summarize, more information = more accuracy. No one needs 144 dB of dynamic range, but that's not the only benefit of using 24-bit. No one needs frequency response up to 96 kHz, but that's not the only benefit of using 192 kHz. Etc.

monoethylene, http://en.wikipedia.org/wiki/Nyquist_frequency. Unless you thought I literally meant "all frequencies," like, all frequencies in existence. I meant, more samples means more accuracy for all representable frequencies relative to the sampling rate. I figured that was implied.

Except that you're not getting more accuracy, you're copying what you do have multiple times. There's not any more there, there if you have a higher sampling rate.

You're talking about upsampling, I'm talking about content natively recorded at a higher sampling rate.

Quote:

Originally Posted by khaos974

But there's point when more improvement and precision is useless, would anyone care if the speedometer of their car was precise to the 1/10000000 miles/hour? Of course not, because they can't perceive that.
When your dithered and noise shaped 16 bit file has a subjective dynamic range of 120 dB, that means the smallest detail is 120 dB softer than the loudest one, isn't that interstitial value already small enough?

I agree with the point that as precision increases, the ability to perceive differences decreases. The question is, what are the limits of human perception? Not easy to answer because not all people are the same and even one person's hearing can fluctuate from day to day depending on weather, allergies, restfulness, etc.

Quote:

Originally Posted by VioletConqueror
The difference is in the exponential growth of binary numbers and in the fact that 16-bit is near perfect while 8-bit is not. 16-bit allows a very large dynamic range with a step size so small that we can't hear it. 8-bit is so much smaller (remember its 2^8 times smaller, not half as small) that to get a reasonable dynamic range you probably need to use a much bigger step-size which sounds worse. Going up to 24-bit doesn't really matter because 16 is already good enough, there is nothing more to gain. It's like refresh rates on a TV, after about 30hz differences are imperceptible because the human optical system doesn't refresh faster than that. Or how a 2Mpixel picture looks exactly the same as a 20Mpixel picture unless you blow it up. Coincidently the advantage of having higher bit-depths is the similar to the advantage of filming well over hi-def 1080 (the red-one from oakley has like 5k resolution I think) that advantage is in processing and filtering. You can use the extra bits/pixels to fiddle with the sound and move the error into the low-order bits so no one notices. You can combine near-by pixels to get a brighter picture or you can use a small portion of the shot for the entire shot etc... In audio there are similar tricks like different filters and such.

I was with you until you said a 30-year-old bit-rate standard is "already good enough." Good enough for whom? For some people, 128 Kbps MP3s are good enough. For some people, vinyl is good enough. However, there is always more to gain, even if not everyone can perceive the difference. Your comment about optical perception works well to illustrate my point. Human sight is not fixed at a certain frequency, it is variable. 24 Hz and 30 Hz are good enough for fluid motion, but they are nowhere near the limits of human perception. Your other point that a 2 MP and a 20 MP picture look the same when the larger image is made smaller does not really apply here; it would be like saying "16-bit and 24-bit sound the same when 24-bit is reduced to 16-bit." When you reduce the detail on purpose, it no longer matters.

You're right, I thought you meant upsampling. But there's still no reason to think that 24 but will give you a better sound.

The limits of human audio perception are well established running from 20 to 20,000 Hz on average with only young children being able to hear the high frequencies and the lower ones generally felt rather than heard. Right now music is sampled at 24 bits, but from everything I've read (there are a couple of articles on this forum that cover it pretty well) what you get is increased dynamic range (headroom), not a better representation of the sound.

I'd be interested in a hypothesis on why more is better, rather than the presumption. The best explanation for why more != better is gregorio's thread from two years ago.